Hi,
I would like to write patch to allow GET_TRANSFERRER_DATA also in pjsip (to
access incoming REFER headers in dialplan). The original patch for chan_sip is
in [1]
Where is the right place to add the code?
I suppose that in res_pjsip_refer.c [2] in function
"refer_incoming_invite_request" I
List
Subject: Re: [asterisk-dev] Gerrit usage
On Thu, Nov 7, 2019 at 8:59 AM Tomec Martin
mailto:to...@ipex.cz>> wrote:
Hi,
after some years I have tried to submit code to gerrit and ended with error:
[asterisk]# git review 13
remote: Resolving deltas: 100% (3/3)
remote: Counting objects:
Hi,
after some years I have tried to submit code to gerrit and ended with error:
[asterisk]# git review 13
remote: Resolving deltas: 100% (3/3)
remote: Counting objects: 66212, done
remote: error: branch refs/publish/13/ASTERISK-28613:
remote: You need 'Create' rights to create new references.
Martin wrote:
> On Fri, Jun 16, 2017, at 11:10 AM, Tomec Martin wrote:
> > Hi,
> > I am looking at issue
> > https://issues.asterisk.org/jira/browse/ASTERISK-27037 and so far I have
> > found that:
> > In asterisk function ast_sip_send_stateful_res
On Fri, Jun 16, 2017, at 11:10 AM, Tomec Martin wrote:
> Hi,
> I am looking at issue
> https://issues.asterisk.org/jira/browse/ASTERISK-27037 and so far I have
> found that:
> In asterisk function ast_sip_send_stateful_response, we receive message
> via pjsip_tsx_recv_msg t
Hi,
I am looking at issue https://issues.asterisk.org/jira/browse/ASTERISK-27037
and so far I have found that:
In asterisk function ast_sip_send_stateful_response, we receive message via
pjsip_tsx_recv_msg then prepare answer and send answer via pjsip_tsx_send_msg.
Before we send the answer, the
9, 2017 at 2:59 AM, Tomec Martin
<to...@ipex.cz<mailto:to...@ipex.cz>> wrote:
So there are 2 ways to move forward:
A) Create RINGNOANSWER event after every call end without answer. That
breaks backward compatibility for thoose who rely on current behavior.
B) Create new eve
Hi all,
I am quite confused about RINGNOANSWER event in queue log. As mentioned in
documentation (https://wiki.asterisk.org/wiki/display/AST/Queue+Logs), this
event should be generated when the call attempt ended without the member
picking up the call.
When the caller hangs up, this event is
On Fri, Dec 9, 2016, at 10:56 AM, Tomec Martin wrote:
> Hi all,
> there are still cases, when member is not removed from pending_members,
> for example in this issue:
> https://issues.asterisk.org/jira/browse/ASTERISK-26621
> There was patch to prevent this (https://gerrit.asterisk
Hi all,
there are still cases, when member is not removed from pending_members, for
example in this issue:
https://issues.asterisk.org/jira/browse/ASTERISK-26621
There was patch to prevent this (https://gerrit.asterisk.org/#/c/3744/1) but it
does not cover all cases.
For now I think that the
-Original Message-
From: asterisk-dev-boun...@lists.digium.com
[mailto:asterisk-dev-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Friday, April 01, 2016 2:01 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] Avoid audio loss in ChanSpy (ref: ASTERISK-25866)
the core debug level to 1, you
will not get as badly flooded as you might if you were to set the level higher.
On Fri, Jan 29, 2016 at 4:15 AM, Tomec Martin
<to...@ipex.cz<mailto:to...@ipex.cz>> wrote:
Hi all,
we are solving issue with mixmonitor – in production asterisk, there are
so
Hi all,
we are solving issue with mixmonitor - in production asterisk, there are
sometimes gaps in recordings (missing syllables), but we cannot reproduce it in
test enviroment.
I suppose it is caused by frames flushing in audiohook.c:
if (ast_test_flag(audiohook,
Hi all,
I am solving issue ASTERISK-19820 and want to write patch, but I am not sure if
I am on good way. Let me explain the issue:
When agent in queue hangs up, his status is updated immediately, but the queue
member lastcall time is updated after a while (after some logging etc.) in
function
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